The present invention relates generally to a noise reduction method.
A frequently used noise reduction method for a disturbed useful signal such as a voice signal, music signal, etc., is spectral subtraction. An advantage of spectral subtraction is the low complexity and that the disturbed useful signal is needed only in one variant (only one channel). A disadvantage consists in the signal delay (caused by the block processing in the spectral domain), the limited maximum attainable noise reduction, and the difficulty in compensating for transient noise. Stationary noise can be reduced, for example, by 12 dB, with the speech still having good quality.
If a higher noise reduction or better speech quality are desired, several recording channels are required. One uses, for example, microphone arrays. Those of the different microphone arrays which make do with small geometrical dimensions for the microphone arrangement are of special interest for many practical applications. Small differential microphone arrays (also referred to as superdirective arrays) are configured as well as an adaptive variant of this microphone arrangement, the LMS (least mean square) algorithm being used for adaptation. In the case of the adaptive form of this array, two microphones are subtracted in two ways with propagation time compensation so as to produce a ‘virtual’ microphone with cardioid or kidney-shaped characteristic toward the speaker and a ‘virtual’ microphone with cardioid characteristic facing away from the speaker. The propagation time compensation corresponds to the time required by the sound for the distance between the two microphones, for example, 1.5 cm. A “back-against-back” cardioid characteristic ensues. The microphone which is directed toward the speaker is the primary signal for the adaptive filter and the microphone directed in the opposite direction is the reference signal of the interference.
FIG. 1 shows an adaptive arrangement for a beam former. The propagation time compensation with an all-pass filter ALL is accomplished by a shift by whole sampled values. The above described combination of two single microphones with omnidirectional characteristic produces a cardioid characteristic toward the speaker and a cardioid characteristic directed in the opposite direction as interference reference. Adaptive filter H1 is adapted in the time domain using the LMS (least mean square) algorithm. A low-pass filter TP at the system output emphasizes low frequency components which are attenuated when the cardioid characteristic is formed.
The tandem arrangement of microphones M according to FIG. 1 is referred to as end fire array whereas the side-by-side arrangement of the microphones is denoted by broadside array.
FIG. 2 shows an arrangement for a broadside array composed of two spaced microphones, the two microphone signals being pre-processed by spectral subtraction (SPS). A propagation time compensation between the two channels is carried out via all-pass filter All and serves to compensate for movements of the speaker. The sum of the two preprocessed microphone signals constitutes the primary input and the difference is the reference input for an adaptive filter H1. The adaptive filter in this arrangement with sum and difference input is also referred to as ‘generalized sidelobe canceller’. The adaptation is carried out using the LMS algorithm, the LMS being implemented in the frequency domain. The microphone signals are post-processed using a modified cross-correlation function in the frequency domain. The fundamental structure including spectral pre-processing via SPS, beam formation, and post-processing (post) is described in European Patent EP 0615226B1, hereby incorporated by reference herein, without exactly specifying the beam former.
FIG. 3 is an overview of microphone circuitry arrangement for the formation of the directivity characteristics for two microphones. The two single microphones themselves can already have a cardioid characteristic or the so-called “omnidirectional characteristic”. “ALL” denotes an all-pass filter for propagation time compensation. ‘Gain’ is a gain compensation between the two channels which is necessary in practice to equalize the sensitivity of the microphone capsules.
The direction of maximum sensitivity in the polar diagrams of the directivity characteristics is 90°. The first 3 arrangements a, b, and c, are suitable as speech channel since a maximum exists at 90° and an attenuation exists for the other directions. Arrangements a and b produce the same directivity characteristic. Arrangements a, b are referred to as sum or difference array and arrangement c is denoted as differential array. Arrangements d and e have a null at 90° in the polar diagram, and are therefore suitable as interference reference. The null at 90° in the polar diagram is necessary to prevent speech components from getting into the reference channel. Speech components in the reference channel lead to partial compensation of speech.
According to arrangements d and e in FIG. 3, a null will occur for the interference reference in the direction toward the speaker under ideal conditions. In practical applications, however, this will not be the case. As a result of this, speech components are treated as interference signals and, consequently, are removed from the actual speech signal.
Beam formers are usually adapted only during speech pauses in order not to permit adaptation to speech components. In this case too, however, speech components present in the reference are compensated for because they are always superimposed on the noise.
Another procedure is to equalize the gain of channels so that, in the ideal case, a null ensues after their subtraction. This is necessary because mass-produced microphones have tolerances. In the arrangements of FIG. 3, this is allowed for by the functional block ‘gain’ which equalizes different microphone sensitivities.
In applications, however, no null is adjusted for the speech signal in the reference in spite of the sensitivity compensation with ‘gain’. Only under the condition that the microphone is operated in the acoustic free-field (without reflections), it is possible for the speech components to be completely compensated for. Real applications have a certain sound component from different directions due to reflections, preventing the occurrence of a null for the speech signal. In the case of arrangements according to FIG. 1 or FIG. 2, a certain speech component will always be found in the reference signal of the beam former, resulting in speech distortions.